• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2132-2135
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• 2003

본 논문에서는 고정소수점 DSP로 구현한 실시간 MP3 오디오 부호화기에 사용되는 초월함수용 하드웨어 가속기 구조를 제안한다. 구현된 하드웨어 가속기는 MP3 부호화 성능을 저하시키는 초월함수 연산오차에 강인하도록 설계되었다. 제안된 가속기의 연산오차는 Q1.23 고정소수점 출력에서 2비트, 즉 2/sup -21/ 까지의 연산오차를 가진다. LAME 부호화기[5]심리음향 모델의 SMR 오차는 테이블 보간법[4]을 사용할 경우에 비해 4dB이상 향상되었으며, 연산량은 총 4 MIPS 감소하였다. 제안한 하드웨어 가속기는 Verilog HDL로 기술되었으며, SYNOPSYS에서 0.18㎛ CMOS 표준 셀 라이브러리 공정으로 합성되었다. 합성 면적은 7514 게이트이며 초월함수 연산에 대한 동작속도는 3 사이클이다.

• The Journal of the Acoustical Society of Korea
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• v.37 no.2
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• pp.110-117
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• 2018

In this paper, we propose a low delay window switching MDCT (Modified Discrete Cosine Transform) method for speech/audio coder. The window switching algorithm is used to reduce the degradation of sound quality in non-stationary trasient duration and to reduce the algorithm delay by using the low delay TDAC (Time Domain Aliasing Cancellation). While the conventional window switching algorithms uses overlap-add with different lengths, the proposed method uses the fixed overlap add length. It results the reduction of algorithm delay by half and 1 bit reduction in frame indication information by using 2 window types. We apply the proposed algorithm to G.729.1 based on MDCT in order to evaluate the performance. The propose method shows the reduction of algorithm delay by half while speech quality of the proposed method maintains same as the conventional method.

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.472-479
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• 2012

This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.

• The Journal of the Acoustical Society of Korea
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• v.17 no.5
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• pp.57-62
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• 1998

본 논문에서는 입력신호의 시변특성에 따라 분석 필터의 대역을 가변 시키는 필터 뱅크의 구조를 제안한다. 제안된 필터뱅크는 일반적으로 32개의 균일한 대역으로 나누어 임 계대역의 표현을 적절히 표현하지 못하는 Polyphase 필터의 단점을 극복하면서 시스템 설 계에 높은 계산량을 요구하는 QMF-tree 필터의 단점을 보완한다. 본 연구에서는 분할 대역 은 4개에서 26개의 대역으로 가변하고, 웨이블릿 필터중 Daubechies필터를 사용하였다. 제 안된 구조의 부호화기는 128kbps에서 MPEG-a오디오와 비슷한 수준의 CD 음질을 유지하 며, 연산량 비교결과는 PolyPhase filter를 이용한 MPEG보다 부호화, 복호화 과정을 합쳐 다양한 전송률과 음원에서 평균 19%의 감소를 얻었다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.149-153
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• 1995

본 논문에서는 TI(Texas Instrument)사의 범용 디지탈 프로세서인 TMS320C30을 이용하여 MPEG-2 계층2(Layer II) 오디오 부호화 알고리듬의 실시간 처리가 가능한 시스템을 구현하였다. 구현한 시스템은 1 채널의 오디오 신호를 처리하기 위한 Slave 보드 5개와 채널 멀티플렉싱과 부가 처리를 위한 Master 보드 1개로 이루어져 있다. MPEG-2 알고리듬의 각 단계별 소요시간을 계산한 후, 이를 바탕으로 각 프로세서에 할당하는 작업량을 조정하여 실시간 처리에 적합한 시스템을 구현하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.770-776
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• 2004

The goal of perceptual audio coder is to reduce redundancy and irrelevancy of audio signal based on the concept of masking. Several studies on masking effect reveal that the masking threshold varies as a function of the noise-like or tone-like nature of audio signals. Therefore, tonality of audio signal influences significantly the quality and efficiency of perceptual audio coder In this paper, we propose a new effective algorithm for tonality measure using spectrum energy. Since the proposed algorithm consists of a few transcendental functions and simple operations, it has lower complexity than MPEG psychoacoustic model-II. The proposed algorithm was tested with some audio signals, and DSP implementation showed that the proposed algorithm could be implemented with 3 MIPS. These results illustrate the efficiency of proposed algorithm in both performance and complexity.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.253-260
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• 2004

This paper proposes a novel algorithm for hybrid lossless audio coding, which employs an integer wavelet transform and a linear prediction model. The proposed algorithm divides the input signal into flames of a proper length, decorrelates the framed data using the integer wavelet transform and linear prediction and finally entropy-codes the frame data. In particular, the adaptive Golomb-Rice coding method used for the entropy coding selects an optimal option which gives the best compression efficiency. Since the proposed algorithm uses integer operations, it significantly improves the computation speed in comparison with an algorithm using real or floating-point operations. When the coding algorithm is implemented in hardware, the system complexity as well as the power consumption is remarkably reduced. Finally, because each frame is independently coded and is byte-aligned with respect to the frame header, it is convenient to move, search, and edit the coded, compressed data.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.261-269
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• 2005

Throughout the history of digital communication, the digital speech coder is used as speech compression tool. Nowadays, the speech coder has been rapidly developed in the area of mobile communication system to overcome severe channel error and limitation of radio frequency resources. Due to the development of high performance communication system, high quality of speech coder is needed. This kind of speech coder can be used not only in communication services but also in digital multimedia services. In this paper, we describe the technologies of digital speech coder which are used in wire and wireless communication. We also present a summary of recent speech coding standards for narrowband and wideband applications. Finally we introduce the technical trends of next generation speech coder.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1
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• pp.18-25
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• 2008

In this paper, new lossless coding method of spectral coefficients for audio codec is proposed. Conventional lossless coder uses Huffman coding utilizing the statistical characteristics of spectral coefficients, but does not provide the high coding efficiency due to its simple structure. To solve this limitation, new lossless coding scheme with better performance is proposed that consists of bit-plane transform and run-length coding. In the proposed scheme, the spectral coefficients are first transformed by bit-plane into 1-D bit-stream with better correlative properties, which is then coded intorun-length and is finally Huffman coded. In addition, the coding performance is further increased by applying the proposed bit-plane coding selectively to each group, after the entire frequency is divided into 3 groups. The performance of proposed coding scheme is measured in terms of theoretical number of bits based on the entropy, and shows at most 6% enhancement compared to that of conventional lossless coder used in AAC audio codec.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.556-562
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• 2013

This paper is aimed to improve performance of Band-Selection speech/audio Coder reconstucted band spectrum that is not sent by the comfort noise. To improve the performance, we use the Spectral Band Replication(SBR) technique instead of substitution of Comfort noise. To synthesize SBR signal, the SBR algorithm is referenced in selected signals and the spectrum synthesized by SBR is injected to non-selected band. Each sub-band spectrum has been energy-weighted by real audio signal. We propose the enhanced the Band-Selection Coder that utilizes synthesized SBR signal from selected signal instead of comfort noise.